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CN101160983B - Method, device and system for data stream processing - Google Patents

Method, device and system for data stream processing Download PDF

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Publication number
CN101160983B
CN101160983B CN2006800127059A CN200680012705A CN101160983B CN 101160983 B CN101160983 B CN 101160983B CN 2006800127059 A CN2006800127059 A CN 2006800127059A CN 200680012705 A CN200680012705 A CN 200680012705A CN 101160983 B CN101160983 B CN 101160983B
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party
coding
decoding algorithm
plug
algorithm plug
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CN101160983A (en
Inventor
蒋砾
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SnapTrack Inc
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Huawei Technologies Co Ltd
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    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04MTELEPHONIC COMMUNICATION
    • H04M7/00Arrangements for interconnection between switching centres
    • H04M7/006Networks other than PSTN/ISDN providing telephone service, e.g. Voice over Internet Protocol (VoIP), including next generation networks with a packet-switched transport layer
    • H04M7/0072Speech codec negotiation
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04LTRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L12/00Data switching networks
    • H04L12/02Details
    • H04L12/16Arrangements for providing special services to substations
    • H04L12/18Arrangements for providing special services to substations for broadcast or conference, e.g. multicast
    • H04L12/1813Arrangements for providing special services to substations for broadcast or conference, e.g. multicast for computer conferences, e.g. chat rooms
    • H04L12/1818Conference organisation arrangements, e.g. handling schedules, setting up parameters needed by nodes to attend a conference, booking network resources, notifying involved parties
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04LTRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/1066Session management
    • H04L65/1069Session establishment or de-establishment
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04LTRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/60Network streaming of media packets
    • H04L65/75Media network packet handling
    • H04L65/762Media network packet handling at the source 
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04LTRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/60Network streaming of media packets
    • H04L65/75Media network packet handling
    • H04L65/764Media network packet handling at the destination 

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  • Engineering & Computer Science (AREA)
  • Multimedia (AREA)
  • Computer Networks & Wireless Communication (AREA)
  • Signal Processing (AREA)
  • General Engineering & Computer Science (AREA)
  • Business, Economics & Management (AREA)
  • General Business, Economics & Management (AREA)
  • Telephonic Communication Services (AREA)

Abstract

The invention provides a method, comprising the following steps: when a calling party and a called party call, sending a coding and decoding algorithm plug-in including a coding and decoding algorithm selected by the calling party or the called party to the other party; and the calling party and the called party encode the data stream transmitted to the opposite party according to the coding and decoding algorithm plug-in and decode the received data stream of the opposite party. The invention makes the communication terminal encode the sent data and decode the received data flow according to the coding and decoding algorithm plug-in, thus realizing intercommunication. Therefore, the number of TCs in the communication link is reduced to the minimum, and the communication quality loss and the time delay caused by the coding and decoding format conversion of the TCs are reduced.

Description

Method, device and system for processing data stream
Technical Field
The present invention relates to the field of communications, and in particular, to a method, an apparatus, and a system for processing data on a communication link.
Background
IP transport mechanisms and TDM transport mechanisms are often employed in communication systems to communicate data streams. Compared with the TDM transmission mechanism, the IP transmission mechanism has the advantages of higher networking flexibility, lower transmission cost, overhead, network construction cost, richer service functions, and the like, and thus becomes a better data transmission mechanism.
In the IP transmission mechanism, as shown in fig. 1, the communication device needs to convert a voice analog signal, i.e. an audio signal, into a digital signal, i.e. an encoded data stream is transmitted over a communication link, and then restore the received digital signal into an audio signal that can be directly received by a user, during which analog/digital and digital/analog conversion needs to be implemented by a codec (TC). The TC of the mobile station is generally set inside the mobile station, the TC of the fixed network terminal is often set inside the terminal or at a media gateway controlled by an access network, and if the communication terminal has a video function, the video stream needs to be encoded and decoded before being displayed on the terminal.
Different terminals (no matter mobile stations or fixed network terminals), different communication network devices (including network elements and servers), different countries, different operators, and different communication systems (such as CDMA2000/WCDMA/TDSCDMA) may use different codec formats, and some common codec formats are as follows:
mu-law Pulse Code Modulation Mu rate
PCMA Pulse Code Modulation, A-law Pulse Code Modulation A rate
13K 13K Vocoder 13K codec
EVRC Enhanced Variable Rate Codec
SMV selective Mode Vocoder
Therefore, it is almost impossible to use the same codec format in the process of transmitting a data stream over a communication link. In order to enable two communication parties to communicate smoothly, TC needs to be inserted into a communication path to realize conversion of different encoding and decoding formats. Communication quality loss and time delay may occur to some extent due to the conversion between different codec formats. Therefore, the fewer TCs for converting different codec formats in the communication link, the better, to ensure higher call quality and smaller transmission delay.
As shown in fig. 2, when the user a and the user B select the same Codec format, for example, the Codec format Codec1 of the user A, B both uses EVRC, thereby realizing interworking; when the user A and the user B adopt different coding and decoding formats, for example, the coding and decoding format Codec1 of the user A adopts EVRC, the coding and decoding format Codec2 of the user B adopts 13K, if TC conversion does not exist in the middle, intercommunication cannot be realized; when TC in the communication path realizes the conversion between Codec1 and Codec2, the intercommunication can be realized.
In general, a plurality of communication nodes passing through the communication link support more than one codec format, and the node a supports three formats, i.e. the codec set supported by the node a is a (a, B, C), the node B supports B (C, d, B), and the node C supports C (e, f, d, g) (the queue already shows the preferred format order of the nodes). The node A carries the A set when sending a 'call request' to the node B; node B sends 'call request' to node C, node C carries B set, node C returns node B with node B and node C intersection (d) through 'call response', because node A does not contain (d) in node B intersection (B, C), AB uses B or C format communication, BC uses d format communication, node B needs TC to make two kinds of communication format conversion, namely when three adjacent nodes supportable format intersection is empty, TC format conversion is necessary to realize intercommunication.
Signalling interaction between the communicating nodes as shown in figure 3,
a. when node a attempts to communicate with node C, node a chooses to pass the voice signal through the intermediate node B. Node A sends a "call request" to node B, which carries Codec1, Codec2 in the Codec format supported by node A.
b. After receiving the 'call request' from the node A, the node B judges that the node B does not support Codec1 but supports Codec3, and the node B sends the 'call request' to the node C, wherein the 'call request' carries Codec2 and Codec 3. The "call request" may also include a called number, a service type, bearer-related parameters, and the like.
c. After node C receives the 'call request' from node B, node C node does not support Codec2, selects a Codec format Codec3 from the 'call request', and returns 'call response' to node B, wherein the Codec format Codec3 selected by node C is carried. During the subsequent call, the Codec3 is used for communication between node B and node C.
d. After node B receives the "Call response" from node C, node B and node C have determined that Codec3 is used, and the "Call request" from node A does not carry Codec3, i.e., node A does not support Codec3, then node B can only select Codec2 to use between node A and node B. The node B returns a "call response" to the node a, which carries the Codec format Codec2 selected by the node B. During the subsequent call, Codec2 is used to communicate between node B and node A. The "call response" may carry the user off-hook indication or by the corresponding subsequent signaling.
Thus there will be a TC at node B to effect a transition between Codec2 and Codec 3.
Sometimes, the selection of the priority of the codec format by the node is different, and the determination can be performed by negotiation with more than one round, and the following description is further given by taking the core network SIP protocol (CDMA2000 and WCDMA are both used) as an example, in conjunction with the figure:
a. as shown in fig. 4, the MSCe1 attempts to initiate a call, and sends an INVITE message to the MSCe2, where SDP1(SDP, Session description protocol, SDP1 is "Session description information 1") is carried. The SDP1 contains a list (a, b, c) of the codec formats supported by MSCe1, the list order indicating the order of selection of the three codec formats by MSCe 1;
MSCe2 returns 183 message to MSCe1, SDP2 carried by said 183 message contains coding and decoding format list (c, b) supported by MSCe2, one is to indicate that the call service is in the process of connection, and the other is to select from the list of SDP1 and send back to MSCe1 the coding and decoding format supported by this end, according to the selection of MSCe2, the format c is prior to the format b;
c. if MSCe1 wishes to select Format b, a Change (UPDATE) (b) message is initiated to MSCe2, carrying SDP3 (b);
MSCe2 returns a200 ok (update) message (SDP3) to MSCe1 indicating agreement to use format b for communication;
msce2 returns a200 ok (invite) message to MSCe 1;
msce1 returns an ACK message to MSCe2 indicating successful receipt of the 200ok (invite) message.
The above-mentioned is 183 unreliable message, and the SIP protocol also defines enhanced 183 reliable message, that is, the following steps are included between the above-mentioned steps b and c:
MSCe1 sends a PRACK (183) message to MSCe2 indicating that MSCe1 successfully received the 183 message;
MSCe2 returns a200 OK (183) message to MSCe1 indicating that mac 2 successfully received the PRACK (183) message.
The defects of the prior art exist in the following aspects:
(1) when the intersection of the codec format sequences respectively supported by two adjacent nodes is empty, even if there is TC, the codec format cannot be converted.
(2) The example of fig. 4 is a two-round questioning of the codec format between the core networks MSCe1 and MSCe2, possibly with more cumbersome questioning and more rounds of confirmation. And similar situations may occur throughout the path including the access network and each of the neighboring network nodes of the core network,
(3) because of more nodes on the communication link, even if the same coding and decoding formats exist between adjacent nodes, the coding and decoding formats can be respectively converted by using a plurality of TCs, and the quality loss and the time delay of voice communication are necessarily generated to a certain extent.
Disclosure of Invention
In view of the above, an object of the present invention is to provide a method and an apparatus for processing a data stream, which can use as few TCs as possible to obtain as high a call quality and as low a time delay as possible.
The data stream processing method provided by the invention comprises the following steps:
when a calling party and a called party call, sending a coding and decoding algorithm plug-in including a coding and decoding algorithm selected by the calling party or the called party to the other party;
and the calling party and the called party encode the data stream transmitted to the opposite party according to the coding and decoding algorithm plug-in and decode the received data stream of the opposite party.
The calling party carries the coding and decoding algorithm plug-in selected by the calling party in the call request and sends the coding and decoding algorithm plug-in to the called party, or the calling party carries the coding and decoding algorithm plug-in set selected by the calling party in the call request and sends the coding and decoding algorithm plug-in to the called party, one coding and decoding algorithm plug-in is selected by the called party, and the selection result is sent to the calling party in the call response message; or the called party carries the coding and decoding algorithm plug-in selected by the called party in the call response and sends the call response to the calling party.
The coding and decoding algorithm plug-in is stored in the terminal equipment of the calling party or in a network server.
The data stream processing device provided by the invention comprises a script compiler and a coder-decoder, wherein:
the script compiler is used for storing the coding and decoding algorithm plug-ins, converting the program scripts of the selected coding and decoding algorithm plug-ins into instruction sequences which can be identified by the processor and sending the instruction sequences to the coder and decoder;
and the coder and the decoder are used for coding the data stream to be transmitted and decoding or coding and decoding format conversion of the received data stream.
The codec is arranged on a communication terminal of a network or a fixed network access gateway.
The network terminal provided by the invention comprises a data stream processing device, wherein the data stream processing device comprises a script compiler and a coder-decoder, and the script compiler comprises:
the script compiler is used for storing the coding and decoding algorithm plug-ins, converting the program scripts of the selected coding and decoding algorithm plug-ins into instruction sequences which can be identified by the processor and sending the instruction sequences to the coder and decoder;
and the coder and the decoder are used for coding the data stream to be transmitted and decoding or coding and decoding format conversion of the received data stream.
The data stream processing system provided by the invention comprises a plurality of network terminals, wherein a calling terminal encodes data to be transmitted through a coding and decoding algorithm plug-in, and a called terminal decodes a received data stream and the calling terminal through the coding and decoding algorithm plug-in.
The coding and decoding algorithm plug-in including the coding and decoding algorithm selected by the calling terminal or the called terminal is sent to the opposite side.
According to the technical scheme provided by the invention, the coding and decoding algorithm plug-in is transparently transmitted to the communication terminal through each node in the communication link, so that the terminal can encode the transmitted data and decode the received data stream according to the coding and decoding algorithm plug-in to realize intercommunication. The number of TCs in the communication link is reduced to the minimum, and the communication quality loss and the time delay caused by the TC for carrying out the coding and decoding format conversion are reduced.
Drawings
FIG. 1 is a functional schematic of a codec (TC);
FIG. 2 is a schematic diagram of the conversion of the codec format implemented by TC;
FIG. 3 is a signaling flow diagram of codec format screening among three communication nodes;
FIG. 4 is a signaling flow diagram of two rounds of codec format negotiation between two communication nodes;
FIG. 5 is a block diagram of a data stream processing apparatus;
FIG. 6 is a signaling flow diagram of obtaining a plug-in for a codec algorithm under one situation in a first method embodiment of the present invention;
fig. 7 is a schematic diagram illustrating a second embodiment of the method of the present invention.
Detailed Description
The invention is described in further detail below with reference to the figures and the embodiments.
Fig. 5 is a data stream processing apparatus including a script interpreter 1 and a codec 2. The script interpreter 1 can store at least one coding and decoding algorithm plug-in, and the coding and decoding algorithm plug-ins respectively correspond to different coding and decoding formats; the script interpreter 1 may also temporarily store the external codec algorithm plug-in, and clear it after the end of a call. The script interpreter 1 analyzes and interprets the algorithm description script in the codec algorithm plug-in, converts the algorithm description script into an instruction sequence which can be recognized by a processor, namely a 'codec program', and transmits the instruction sequence to the codec2 for loading. It should be noted that the codec algorithm plug-in stored in the script interpreter 1 may also be stored in another network server and downloaded when necessary.
The codec2, after being loaded with the codec program, can encode, decode, and format-convert the data stream.
The data stream processing device shown in fig. 5 is arranged on a mobile station or a fixed network terminal, or the data stream processing device is arranged at a fixed network access network gateway, so that coding can be carried out while a voice analog signal is converted into a digital audio signal, or decoding can be carried out while the digital audio signal is converted into the voice analog signal, and coding and decoding can be carried out before and after the video signal is transmitted on a channel. The data stream processing apparatus shown in fig. 5 is arranged in a conference processing unit at a media gateway of a main control party during a multi-party call, i.e., a conference call, and decoding is performed before multi-party sound mixing, and encoding is performed after multi-party sound mixing, so that transmission of a multi-path signal to a single-path signal can be realized, and the multi-party call is realized.
The following is an embodiment of the method for implementing data stream processing of the present invention:
in the first embodiment, a calling party initiates a call request to indicate that both parties obtain a codec algorithm plug-in during call setup. The following can be taken into account depending on the supply source of the plug-in:
a. optionally, codec algorithm plug-ins with different formats are stored in a device of a calling party, that is, the data stream processing apparatus shown in fig. 5 is arranged in the device of the calling party, a data stream processing flow of this embodiment is shown in fig. 6, the calling party selects a codec algorithm plug-in with codec format 1, the plug-in is carried when sending a call Request to a BSC, and the plug-in is transmitted to an opposite terminal through each network node of an access network or a core network along with a signaling, for example, in an access network part, a bearer parameter carried by the BSC in a CM (CM, Connection Management) Service Request (CM Service Request) message sent to the MSC or an Assignment Request (Assignment Request) message sent by the MSC to the BSC includes the codec algorithm plug-in; at the core network part, MSCe1 carries the plug-in a call request (INVITE) message to MSCe 2. In this figure, the specific node names are omitted for simplicity and all nodes are replaced with intermediate nodes, i.e., communication node a. The other terminal, called party, temporarily stores the plug-in, and in the call, the calling party and the called party encode the data stream to be transmitted by using the same encoding and decoding format of the encoding and decoding algorithm plug-in and decode the received data stream in the full duplex mode. In this way, data is transmitted directly over the communication link without conversion of the data format. When the call is finished, the called terminal can release the plug-in or store the plug-in as the own optional plug-in of the coding and decoding algorithm.
b. The optional plug-in for codec algorithms of different formats is stored in a server of the communication network, that is, the script interpreter in the data stream processing apparatus shown in fig. 5 is set in the server of the communication network, and the codec is set in the device of the calling party and carried by the communication signaling (such as call response) to the calling party, and the calling party selects one of the two and transmits the selected one to the called party through the signaling, so that the calling party and the called party can encode and decode the data stream in the codec format of the same plug-in for codec algorithms in a full duplex manner.
In the second embodiment, the calling party initiates a call request and specifies that the called party determines the codec algorithm plug-in, and in this embodiment, the following three processing modes may be adopted according to different sources of the codec algorithm plug-in:
a. a set of codec algorithm plug-ins that have been selected by itself may be delivered by the calling party for selection by the called party;
b. directly appointing the own coding and decoding algorithm plug-in by the called party; or,
c. when the optional codec algorithm plug-ins for different formats are in a server of the communication network, the codec algorithm plug-in is selected by the called party from the plug-ins provided by the server.
The called party returns the selected coding and decoding algorithm plug-in to the calling party through the response message, and the calling party and the called party encode and decode the data stream in the coding and decoding format of the same coding and decoding algorithm plug-in a full duplex mode.
Example three: in the process of call connection between the calling party and the called party according to the common connection process, one party provides a coding and decoding algorithm plug-in after successful negotiation, once the call connection is established, the provider immediately sends the coding and decoding algorithm plug-in selected by the provider, and the calling party and the called party encode and decode data streams in the same coding and decoding format in a full duplex mode.
It should be noted that, when one party in communication is a fixed network terminal and the codec in the data stream processing apparatus shown in fig. 5 is installed in a media gateway controlled by an access network, the codec algorithm plug-in is loaded on the codec of the media gateway,
the above embodiments do not exhaust the situation that the calling and called parties negotiate in the process of establishing the call connection so as to obtain the determined codec algorithm plug-in used for communication, and it should be regarded that any mode is within the protection scope of the present invention.
In a multi-party call of three or more parties, for example, in a call scenario, since both parties of an initial call have performed coding and decoding conversion in a certain same coding and decoding format using a coding and decoding algorithm plug-in, a conference processing unit of a media gateway of a main control party stores the coding and decoding algorithm plug-in, and in the process that a late joining party obtains access permission, the plug-in is transmitted to the late joining party along with signaling, and a data stream is also processed at the later joining party. Setting A, B, C three parties to carry out conference communication, respectively decoding data streams transmitted from A, B by a coding and decoding algorithm plug-in on a conference processing unit, then carrying out sound mixing processing, coding into a data stream, and transmitting the data stream to C; after receiving the data stream, the terminal C decodes the data stream to simultaneously receive the voice or video signals of a and B.
It should also be noted that, when one party in communication is a fixed network terminal and the codec in the data stream processing apparatus shown in fig. 5 is installed in a media gateway controlled by an access network, the codec algorithm plug-in is loaded on the TC of the media gateway,
the above description is only for the preferred embodiment of the present invention, and is not intended to limit the scope of the present invention. Any modification, equivalent replacement, improvement and the like made within the spirit and principle of the present invention are included in the scope of the present invention.

Claims (4)

1. A method of processing a data stream, characterized in that,
when a conference call is carried out, a conference processing unit of a calling party media gateway saves an encoding and decoding algorithm plug-in at the conversation stage of an initial conversation party and a calling party; when a calling party and a called party are calling, a coding and decoding algorithm plug-in selected by the calling party or the called party and comprising a coding and decoding algorithm is sent to the other party, and the conference processing unit decodes data streams of two or more parties according to the coding and decoding algorithm plug-in, performs coding after mixing the data streams, and then transmits the data streams to conference participants of the other party;
and the calling party and the called party encode the data stream transmitted to the opposite party according to the coding and decoding algorithm plug-in and decode the received data stream of the opposite party.
2. The method of claim 1, wherein the calling party carries the codec algorithm plug-in selected by itself in the call request and sends the call request to the called party; or the calling party carries the coding and decoding algorithm plug-in assembly selected by the calling party in the call request and sends the coding and decoding algorithm plug-in assembly to the called party, the called party selects one coding and decoding algorithm plug-in assembly, and the selection result is sent to the calling party in the call response message; or the called party carries the coding and decoding algorithm plug-in selected by the called party in the call response and sends the call response to the calling party.
3. Method according to claim 1 or 2, characterized in that the codec algorithm plug-in is stored in the terminal device of the calling party or in a network server.
4. The method of claim 1, wherein a subsequent party participating in a conference call signals a codec algorithm plug-in from the conference processing unit; or a codec plug-in is fetched from the conference processing unit as soon as a call connection is established.
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Families Citing this family (8)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US10038760B2 (en) * 2009-06-01 2018-07-31 International Business Machines Corporation System and method to support codec delivery models for IMS networks
CN101719370A (en) * 2009-11-25 2010-06-02 中兴通讯股份有限公司 Device and method for realizing reconfiguration of mobile terminal audio encoding-decoding algorithm
US9203633B2 (en) 2011-10-27 2015-12-01 Polycom, Inc. Mobile group conferencing with portable devices
CN106250556B (en) * 2016-08-17 2019-06-18 贵州数据宝网络科技有限公司 Data Mining Methods for Big Data Analysis
WO2020101776A2 (en) * 2018-08-13 2020-05-22 Applied Avionics, Inc. Command interpreter or command parser based control architecture for aircraft control, interface units and/or illuminated pushbutton switches
CN110009463B (en) * 2018-11-20 2023-08-04 创新先进技术有限公司 Data communication processing system and method
CN112291568B (en) * 2020-11-13 2022-12-20 Oppo广东移动通信有限公司 Data processing method, device, medium, network access equipment and electronic equipment
CN114501149B (en) * 2022-02-10 2024-09-24 天脉拓道(北京)科技有限公司 Audio/video file decoding method, device, equipment and readable medium

Citations (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN1250327A (en) * 1998-08-18 2000-04-12 日本电气株式会社 Connection among moving tables with different code rules
CN1435045A (en) * 1999-12-16 2003-08-06 艾利森电话股份有限公司 Method for changing quality of service for voice over IP calls
WO2004054286A2 (en) * 2002-12-06 2004-06-24 Qualcomm, Incorporated Techniques for supporting gsm to w-cdma reselection

Family Cites Families (9)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US5838664A (en) * 1997-07-17 1998-11-17 Videoserver, Inc. Video teleconferencing system with digital transcoding
US6185259B1 (en) * 1996-06-12 2001-02-06 Ericsson Inc. Transmitter/receiver for GMSK and offset-QAM
US6603774B1 (en) * 1998-10-09 2003-08-05 Cisco Technology, Inc. Signaling and handling method for proxy transcoding of encoded voice packets in packet telephony applications
ES2353854T3 (en) * 2000-08-14 2011-03-07 Nokia Siemens Networks Oy COMMUNICATION SYSTEM AND METHOD TO PROVIDE A MODE SELECTION PROCEDURE.
US20030014488A1 (en) * 2001-06-13 2003-01-16 Siddhartha Dalal System and method for enabling multimedia conferencing services on a real-time communications platform
US7406096B2 (en) * 2002-12-06 2008-07-29 Qualcomm Incorporated Tandem-free intersystem voice communication
KR100591890B1 (en) * 2003-04-01 2006-06-20 한국전자통신연구원 Adaptive Transceiver Method and Apparatus in Multi Antenna Wireless Communication System
US7149515B2 (en) * 2003-10-17 2006-12-12 Motorola, Inc. Vocoder selection method
US20060041871A1 (en) * 2004-04-27 2006-02-23 Richard Friedman Resource description framework transcoder repository and methods for exposing data assets

Patent Citations (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN1250327A (en) * 1998-08-18 2000-04-12 日本电气株式会社 Connection among moving tables with different code rules
CN1435045A (en) * 1999-12-16 2003-08-06 艾利森电话股份有限公司 Method for changing quality of service for voice over IP calls
WO2004054286A2 (en) * 2002-12-06 2004-06-24 Qualcomm, Incorporated Techniques for supporting gsm to w-cdma reselection

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